X-Git-Url: http://www.chiark.greenend.org.uk/ucgi/~mdw/git/disorder/blobdiff_plain/e83d0967d4c0965eb8036248acc20d1bf12ad1d8..af66d051899a3ac7d37095807ff14069f0815285:/clients/playrtp.c diff --git a/clients/playrtp.c b/clients/playrtp.c index 7ec35d0..9757be6 100644 --- a/clients/playrtp.c +++ b/clients/playrtp.c @@ -17,6 +17,37 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 * USA */ +/** @file clients/playrtp.c + * @brief RTP player + * + * This player supports Linux (ALSA) + * and Apple Mac (Core Audio) + * systems. There is no support for Microsoft Windows yet, and that will in + * fact probably an entirely separate program. + * + * The program runs (at least) three threads. listen_thread() is responsible + * for reading RTP packets off the wire and adding them to the linked list @ref + * received_packets, assuming they are basically sound. queue_thread() takes + * packets off this linked list and adds them to @ref packets (an operation + * which might be much slower due to contention for @ref lock). + * + * The main thread is responsible for actually playing audio. In ALSA this + * means it waits until ALSA says it's ready for more audio which it then + * plays. See @ref clients/playrtp-alsa.c. + * + * In Core Audio the main thread is only responsible for starting and stopping + * play: the system does the actual playback in its own private thread, and + * calls adioproc() to fetch the audio data. See @ref + * clients/playrtp-coreaudio.c. + * + * Sometimes it happens that there is no audio available to play. This may + * because the server went away, or a packet was dropped, or the server + * deliberately did not send any sound because it encountered a silence. + * + * Assumptions: + * - it is safe to read uint32_t values without a lock protecting them + */ #include #include "types.h" @@ -29,6 +60,15 @@ #include #include #include +#include +#include +#include +#include +#include +#include +#include +#include +#include #include "log.h" #include "mem.h" @@ -36,68 +76,289 @@ #include "addr.h" #include "syscalls.h" #include "rtp.h" -#include "debug.h" +#include "defs.h" +#include "vector.h" +#include "heap.h" +#include "timeval.h" +#include "client.h" +#include "playrtp.h" +#include "inputline.h" -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -# include +#define readahead linux_headers_are_borked + +/** @brief Obsolete synonym */ +#ifndef IPV6_JOIN_GROUP +# define IPV6_JOIN_GROUP IPV6_ADD_MEMBERSHIP #endif +/** @brief RTP socket */ static int rtpfd; -#define MAXSAMPLES 2048 /* max samples/frame we'll support */ -/* NB two channels = two samples in this program! */ -#define MINBUFFER 8820 /* when to stop playing */ -#define READAHEAD 88200 /* how far to read ahead */ -#define MAXBUFFER (3 * 88200) /* maximum buffer contents */ - -struct frame { - struct frame *next; /* another frame */ - int nsamples; /* number of samples */ - int nused; /* number of samples used so far */ - uint32_t timestamp; /* timestamp from packet */ -#if HAVE_COREAUDIO_AUDIOHARDWARE_H - float samples[MAXSAMPLES]; /* converted sample data */ +/** @brief Log output */ +static FILE *logfp; + +/** @brief Output device */ +const char *device; + +/** @brief Minimum low watermark + * + * We'll stop playing if there's only this many samples in the buffer. */ +unsigned minbuffer = 2 * 44100 / 10; /* 0.2 seconds */ + +/** @brief Buffer high watermark + * + * We'll only start playing when this many samples are available. */ +static unsigned readahead = 2 * 2 * 44100; + +/** @brief Maximum buffer size + * + * We'll stop reading from the network if we have this many samples. */ +static unsigned maxbuffer; + +/** @brief Received packets + * Protected by @ref receive_lock + * + * Received packets are added to this list, and queue_thread() picks them off + * it and adds them to @ref packets. Whenever a packet is added to it, @ref + * receive_cond is signalled. + */ +struct packet *received_packets; + +/** @brief Tail of @ref received_packets + * Protected by @ref receive_lock + */ +struct packet **received_tail = &received_packets; + +/** @brief Lock protecting @ref received_packets + * + * Only listen_thread() and queue_thread() ever hold this lock. It is vital + * that queue_thread() not hold it any longer than it strictly has to. */ +pthread_mutex_t receive_lock = PTHREAD_MUTEX_INITIALIZER; + +/** @brief Condition variable signalled when @ref received_packets is updated + * + * Used by listen_thread() to notify queue_thread() that it has added another + * packet to @ref received_packets. */ +pthread_cond_t receive_cond = PTHREAD_COND_INITIALIZER; + +/** @brief Length of @ref received_packets */ +uint32_t nreceived; + +/** @brief Binary heap of received packets */ +struct pheap packets; + +/** @brief Total number of samples available + * + * We make this volatile because we inspect it without a protecting lock, + * so the usual pthread_* guarantees aren't available. + */ +volatile uint32_t nsamples; + +/** @brief Timestamp of next packet to play. + * + * This is set to the timestamp of the last packet, plus the number of + * samples it contained. Only valid if @ref active is nonzero. + */ +uint32_t next_timestamp; + +/** @brief True if actively playing + * + * This is true when playing and false when just buffering. */ +int active; + +/** @brief Lock protecting @ref packets */ +pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; + +/** @brief Condition variable signalled whenever @ref packets is changed */ +pthread_cond_t cond = PTHREAD_COND_INITIALIZER; + +#if HAVE_ALSA_ASOUNDLIB_H +# define DEFAULT_BACKEND playrtp_alsa +#elif HAVE_SYS_SOUNDCARD_H || EMPEG_HOST +# define DEFAULT_BACKEND playrtp_oss +#elif HAVE_COREAUDIO_AUDIOHARDWARE_H +# define DEFAULT_BACKEND playrtp_coreaudio +#else +# error No known backend #endif -}; -static unsigned long nsamples; /* total samples available */ +/** @brief Backend to play with */ +static void (*backend)(void) = &DEFAULT_BACKEND; -static struct frame *frames; /* received frames in ascending order - * of timestamp */ -static pthread_mutex_t lock = PTHREAD_MUTEX_INITIALIZER; -/* lock protecting frame list */ +HEAP_DEFINE(pheap, struct packet *, lt_packet); -static pthread_cond_t cond = PTHREAD_CONDVAR_INITIALIZER; -/* signalled whenever we add a new frame */ +/** @brief Control socket or NULL */ +const char *control_socket; static const struct option options[] = { { "help", no_argument, 0, 'h' }, { "version", no_argument, 0, 'V' }, { "debug", no_argument, 0, 'd' }, + { "device", required_argument, 0, 'D' }, + { "min", required_argument, 0, 'm' }, + { "max", required_argument, 0, 'x' }, + { "buffer", required_argument, 0, 'b' }, + { "rcvbuf", required_argument, 0, 'R' }, +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + { "oss", no_argument, 0, 'o' }, +#endif +#if HAVE_ALSA_ASOUNDLIB_H + { "alsa", no_argument, 0, 'a' }, +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + { "core-audio", no_argument, 0, 'c' }, +#endif + { "socket", required_argument, 0, 's' }, + { "config", required_argument, 0, 'C' }, { 0, 0, 0, 0 } }; -/* Return true iff a > b in sequence-space arithmetic */ -static inline int gt(const struct frame *a, const struct frame *b) { - return (uint32_t)(a->timestamp - b->timestamp) < 0x80000000; +/** @brief Control thread + * + * This thread is responsible for accepting control commands from Disobedience + * (or other controllers) over an AF_UNIX stream socket with a path specified + * by the @c --socket option. The protocol uses simple string commands and + * replies: + * + * - @c stop will shut the player down + * - @c query will send back the reply @c running + * - anything else is ignored + * + * Commands and response strings terminated by shutting down the connection or + * by a newline. No attempt is made to multiplex multiple clients so it is + * important that the command be sent as soon as the connection is made - it is + * assumed that both parties to the protocol are entirely cooperating with one + * another. + */ +static void *control_thread(void attribute((unused)) *arg) { + struct sockaddr_un sa; + int sfd, cfd; + char *line; + socklen_t salen; + FILE *fp; + + assert(control_socket); + unlink(control_socket); + memset(&sa, 0, sizeof sa); + sa.sun_family = AF_UNIX; + strcpy(sa.sun_path, control_socket); + sfd = xsocket(PF_UNIX, SOCK_STREAM, 0); + if(bind(sfd, (const struct sockaddr *)&sa, sizeof sa) < 0) + fatal(errno, "error binding to %s", control_socket); + if(listen(sfd, 128) < 0) + fatal(errno, "error calling listen on %s", control_socket); + info("listening on %s", control_socket); + for(;;) { + salen = sizeof sa; + cfd = accept(sfd, (struct sockaddr *)&sa, &salen); + if(cfd < 0) { + switch(errno) { + case EINTR: + case EAGAIN: + break; + default: + fatal(errno, "error calling accept on %s", control_socket); + } + } + if(!(fp = fdopen(cfd, "r+"))) { + error(errno, "error calling fdopen for %s connection", control_socket); + close(cfd); + continue; + } + if(!inputline(control_socket, fp, &line, '\n')) { + if(!strcmp(line, "stop")) { + info("stopped via %s", control_socket); + exit(0); /* terminate immediately */ + } + if(!strcmp(line, "query")) + fprintf(fp, "running"); + xfree(line); + } + if(fclose(fp) < 0) + error(errno, "error closing %s connection", control_socket); + } +} + +/** @brief Drop the first packet + * + * Assumes that @ref lock is held. + */ +static void drop_first_packet(void) { + if(pheap_count(&packets)) { + struct packet *const p = pheap_remove(&packets); + nsamples -= p->nsamples; + playrtp_free_packet(p); + pthread_cond_broadcast(&cond); + } +} + +/** @brief Background thread adding packets to heap + * + * This just transfers packets from @ref received_packets to @ref packets. It + * is important that it holds @ref receive_lock for as little time as possible, + * in order to minimize the interval between calls to read() in + * listen_thread(). + */ +static void *queue_thread(void attribute((unused)) *arg) { + struct packet *p; + + for(;;) { + /* Get the next packet */ + pthread_mutex_lock(&receive_lock); + while(!received_packets) + pthread_cond_wait(&receive_cond, &receive_lock); + p = received_packets; + received_packets = p->next; + if(!received_packets) + received_tail = &received_packets; + --nreceived; + pthread_mutex_unlock(&receive_lock); + /* Add it to the heap */ + pthread_mutex_lock(&lock); + pheap_insert(&packets, p); + nsamples += p->nsamples; + pthread_cond_broadcast(&cond); + pthread_mutex_unlock(&lock); + } } -/* Background thread that reads frames over the network and add them to the - * list */ -static listen_thread(void attribute((unused)) *arg) { - struct frame *f = 0, **ff; - int n, i; - union { - struct rtp_header header; - uint8_t bytes[sizeof(uint16_t) * MAXSAMPLES + sizeof (struct rtp_header)]; - } packet; - const uint16_t *const samples = (uint16_t *)(packet.bytes - + sizeof (struct rtp_header)); +/** @brief Background thread collecting samples + * + * This function collects samples, perhaps converts them to the target format, + * and adds them to the packet list. + * + * It is crucial that the gap between successive calls to read() is as small as + * possible: otherwise packets will be dropped. + * + * We use a binary heap to ensure that the unavoidable effort is at worst + * logarithmic in the total number of packets - in fact if packets are mostly + * received in order then we will largely do constant work per packet since the + * newest packet will always be last. + * + * Of more concern is that we must acquire the lock on the heap to add a packet + * to it. If this proves a problem in practice then the answer would be + * (probably doubly) linked list with new packets added the end and a second + * thread which reads packets off the list and adds them to the heap. + * + * We keep memory allocation (mostly) very fast by keeping pre-allocated + * packets around; see @ref playrtp_new_packet(). + */ +static void *listen_thread(void attribute((unused)) *arg) { + struct packet *p = 0; + int n; + struct rtp_header header; + uint16_t seq; + uint32_t timestamp; + struct iovec iov[2]; for(;;) { - if(!f) - f = xmalloc(sizeof *f); - n = read(rtpfd, packet.bytes, sizeof packet.bytes); + if(!p) + p = playrtp_new_packet(); + iov[0].iov_base = &header; + iov[0].iov_len = sizeof header; + iov[1].iov_base = p->samples_raw; + iov[1].iov_len = sizeof p->samples_raw / sizeof *p->samples_raw; + n = readv(rtpfd, iov, 2); if(n < 0) { switch(errno) { case EINTR: @@ -106,153 +367,117 @@ static listen_thread(void attribute((unused)) *arg) { fatal(errno, "error reading from socket"); } } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H + /* Ignore too-short packets */ + if((size_t)n <= sizeof (struct rtp_header)) { + info("ignored a short packet"); + continue; + } + timestamp = htonl(header.timestamp); + seq = htons(header.seq); + /* Ignore packets in the past */ + if(active && lt(timestamp, next_timestamp)) { + info("dropping old packet, timestamp=%"PRIx32" < %"PRIx32, + timestamp, next_timestamp); + continue; + } + p->next = 0; + p->flags = 0; + p->timestamp = timestamp; /* Convert to target format */ - switch(packet.header.mtp & 0x7F) { + if(header.mpt & 0x80) + p->flags |= IDLE; + switch(header.mpt & 0x7F) { case 10: - f->nsamples = (n - sizeof (struct rtp_header)) / sizeof(uint16_t); - for(i = 0; i < f->nsamples; ++i) - f->samples[i] = (int16_t)ntohs(samples[i]) * (0.5f / 32767); + p->nsamples = (n - sizeof header) / sizeof(uint16_t); break; /* TODO support other RFC3551 media types (when the speaker does) */ default: - fatal(0, "unsupported RTP payload type %d", - packet.header.mpt & 0x7F); + fatal(0, "unsupported RTP payload type %d", + header.mpt & 0x7F); } -#endif - f->used = 0; - f->timestamp = ntohl(packet.header.timestamp); - pthread_mutex_lock(&lock); - /* Stop reading if we've reached the maximum */ - while(nsamples >= MAXBUFFER) - pthread_cond_wait(&cond, &lock); - for(ff = &frames; *ff && !gt(*ff, f); ff = &(*ff)->next) - ; - f->next = *ff; - *ff = f; - nsamples += f->nsamples; - pthread_cond_broadcast(&cond); - pthread_mutex_unlock(&lock); - f = 0; + if(logfp) + fprintf(logfp, "sequence %u timestamp %"PRIx32" length %"PRIx32" end %"PRIx32"\n", + seq, timestamp, p->nsamples, timestamp + p->nsamples); + /* Stop reading if we've reached the maximum. + * + * This is rather unsatisfactory: it means that if packets get heavily + * out of order then we guarantee dropouts. But for now... */ + if(nsamples >= maxbuffer) { + pthread_mutex_lock(&lock); + while(nsamples >= maxbuffer) + pthread_cond_wait(&cond, &lock); + pthread_mutex_unlock(&lock); + } + /* Add the packet to the receive queue */ + pthread_mutex_lock(&receive_lock); + *received_tail = p; + received_tail = &p->next; + ++nreceived; + pthread_cond_signal(&receive_cond); + pthread_mutex_unlock(&receive_lock); + /* We'll need a new packet */ + p = 0; } } -#if HAVE_COREAUDIO_AUDIOHARDWARE_H -static OSStatus adioproc(AudioDeviceID inDevice, - const AudioTimeStamp *inNow, - const AudioBufferList *inInputData, - const AudioTimeStamp *inInputTime, - AudioBufferList *outOutputData, - const AudioTimeStamp *inOutputTime, - void *inClientData) { - UInt32 nbuffers = outOutputData->mNumberBuffers; - AudioBuffer *ab = outOutputData->mBuffers; - float *samplesOut; /* where to write samples to */ - size_t samplesOutLeft; /* space left */ - size_t samplesInLeft; - size_t samplesToCopy; - - pthread_mutex_lock(&lock); - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); - while(frames && nbuffers > 0) { - if(frames->used == frames->nsamples) { - /* TODO if we dropped a packet then we should introduce a gap here */ - struct frame *const f = frames; - frames = f->next; - free(f); - pthread_cond_broadcast(&cond); - continue; - } - if(samplesOutLeft == 0) { - --nbuffers; - ++ab; - samplesOut = ab->data; - samplesOutLeft = ab->mDataByteSize / sizeof (float); - continue; - } - /* Now: (1) there is some data left to read - * (2) there is some space to put it */ - samplesInLeft = frames->nsamples - frames->used; - samplesToCopy = (samplesInLeft < samplesOutLeft - ? samplesInLeft : samplesOutLeft); - memcpy(samplesOut, frame->samples + frames->used, samplesToCopy); - frames->used += samplesToCopy; - samplesOut += samplesToCopy; - samesOutLeft -= samplesToCopy; +/** @brief Wait until the buffer is adequately full + * + * Must be called with @ref lock held. + */ +void playrtp_fill_buffer(void) { + while(nsamples) + drop_first_packet(); + info("Buffering..."); + while(nsamples < readahead) + pthread_cond_wait(&cond, &lock); + next_timestamp = pheap_first(&packets)->timestamp; + active = 1; +} + +/** @brief Find next packet + * @return Packet to play or NULL if none found + * + * The return packet is merely guaranteed not to be in the past: it might be + * the first packet in the future rather than one that is actually suitable to + * play. + * + * Must be called with @ref lock held. + */ +struct packet *playrtp_next_packet(void) { + while(pheap_count(&packets)) { + struct packet *const p = pheap_first(&packets); + if(le(p->timestamp + p->nsamples, next_timestamp)) { + /* This packet is in the past. Drop it and try another one. */ + drop_first_packet(); + } else + /* This packet is NOT in the past. (It might be in the future + * however.) */ + return p; } - pthread_mutex_unlock(&lock); return 0; } -#endif -void play_rtp(void) { - pthread_t lt; +/** @brief Play an RTP stream + * + * This is the guts of the program. It is responsible for: + * - starting the listening thread + * - opening the audio device + * - reading ahead to build up a buffer + * - arranging for audio to be played + * - detecting when the buffer has got too small and re-buffering + */ +static void play_rtp(void) { + pthread_t ltid; + int err; /* We receive and convert audio data in a background thread */ - pthread_create(<, 0, listen_thread, 0); -#if API_ALSA - assert(!"implemented"); -#elif HAVE_COREAUDIO_AUDIOHARDWARE_H - { - OSStatus status; - UInt32 propertySize; - AudioDeviceID adid; - AudioStreamBasicDescription asbd; - - /* If this looks suspiciously like libao's macosx driver there's an - * excellent reason for that... */ - - /* TODO report errors as strings not numbers */ - propertySize = sizeof adid; - status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, - &propertySize, &adid); - if(status) - fatal(0, "AudioHardwareGetProperty: %d", (int)status); - if(adid == kAudioDeviceUnknown) - fatal(0, "no output device"); - propertySize = sizeof asbd; - status = AudioDeviceGetProperty(adid, 0, false, - kAudioDevicePropertyStreamFormat, - &propertySize, &asbd); - if(status) - fatal(0, "AudioHardwareGetProperty: %d", (int)status); - D(("mSampleRate %f", asbd.mSampleRate)); - D(("mFormatID %08"PRIx32, asbd.mFormatID)); - D(("mFormatFlags %08"PRIx32, asbd.mFormatFlags)); - D(("mBytesPerPacket %08"PRIx32, asbd.mBytesPerPacket)); - D(("mFramesPerPacket %08"PRIx32, asbd.mFramesPerPacket)); - D(("mBytesPerFrame %08"PRIx32, asbd.mBytesPerFrame)); - D(("mChannelsPerFrame %08"PRIx32, asbd.mChannelsPerFrame)); - D(("mBitsPerChannel %08"PRIx32, asbd.mBitsPerChannel)); - D(("mReserved %08"PRIx32, asbd.mReserved)); - if(asbd.mFormatID != kAudioFormatLinearPCM) - fatal(0, "audio device does not support kAudioFormatLinearPCM"); - status = AudioDeviceAddIOProc(adid, adioproc, 0); - if(status) - fatal(0, "AudioDeviceAddIOProc: %d", (int)status); - pthread_mutex_lock(&lock); - for(;;) { - /* Wait for the buffer to fill up a bit */ - while(nsamples < READAHEAD) - pthread_cond_wait(&cond, &lock); - /* Start playing now */ - status = AudioDeviceStart(adid, adioproc); - if(status) - fatal(0, "AudioDeviceStart: %d", (int)status); - /* Wait until the buffer empties out */ - while(nsamples >= MINBUFFER) - pthread_cond_wait(&cond, &lock); - /* Stop playing for a bit until the buffer re-fills */ - status = AudioDeviceStop(adid, adioproc); - if(status) - fatal(0, "AudioDeviceStop: %d", (int)status); - /* Go back round */ - } - } -#else -# error No known audio API -#endif + if((err = pthread_create(<id, 0, listen_thread, 0))) + fatal(err, "pthread_create listen_thread"); + /* We have a second thread to add received packets to the queue */ + if((err = pthread_create(<id, 0, queue_thread, 0))) + fatal(err, "pthread_create queue_thread"); + /* The rest of the work is backend-specific */ + backend(); } /* display usage message and terminate */ @@ -260,9 +485,24 @@ static void help(void) { xprintf("Usage:\n" " disorder-playrtp [OPTIONS] ADDRESS [PORT]\n" "Options:\n" + " --device, -D DEVICE Output device\n" + " --min, -m FRAMES Buffer low water mark\n" + " --buffer, -b FRAMES Buffer high water mark\n" + " --max, -x FRAMES Buffer maximum size\n" + " --rcvbuf, -R BYTES Socket receive buffer size\n" + " --config, -C PATH Set configuration file\n" +#if HAVE_ALSA_ASOUNDLIB_H + " --alsa, -a Use ALSA to play audio\n" +#endif +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + " --oss, -o Use OSS to play audio\n" +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + " --core-audio, -c Use Core Audio to play audio\n" +#endif " --help, -h Display usage message\n" " --version, -V Display version number\n" - " --debug, -d Turn on debugging\n"); + ); xfclose(stdout); exit(0); } @@ -275,12 +515,25 @@ static void version(void) { } int main(int argc, char **argv) { - int n; + int n, err; struct addrinfo *res; struct stringlist sl; - const char *sockname; + char *sockname; + int rcvbuf, target_rcvbuf = 131072; + socklen_t len; + struct ip_mreq mreq; + struct ipv6_mreq mreq6; + disorder_client *c; + char *address, *port; + int is_multicast; + union any_sockaddr { + struct sockaddr sa; + struct sockaddr_in in; + struct sockaddr_in6 in6; + }; + union any_sockaddr mgroup; - static const struct addrinfo prefbind = { + static const struct addrinfo prefs = { AI_PASSIVE, PF_INET, SOCK_DGRAM, @@ -293,29 +546,136 @@ int main(int argc, char **argv) { mem_init(); if(!setlocale(LC_CTYPE, "")) fatal(errno, "error calling setlocale"); - while((n = getopt_long(argc, argv, "hVd", options, 0)) >= 0) { + while((n = getopt_long(argc, argv, "hVdD:m:b:x:L:R:M:aocC:", options, 0)) >= 0) { switch(n) { case 'h': help(); case 'V': version(); case 'd': debugging = 1; break; + case 'D': device = optarg; break; + case 'm': minbuffer = 2 * atol(optarg); break; + case 'b': readahead = 2 * atol(optarg); break; + case 'x': maxbuffer = 2 * atol(optarg); break; + case 'L': logfp = fopen(optarg, "w"); break; + case 'R': target_rcvbuf = atoi(optarg); break; +#if HAVE_ALSA_ASOUNDLIB_H + case 'a': backend = playrtp_alsa; break; +#endif +#if HAVE_SYS_SOUNDCARD_H || EMPEG_HOST + case 'o': backend = playrtp_oss; break; +#endif +#if HAVE_COREAUDIO_AUDIOHARDWARE_H + case 'c': backend = playrtp_coreaudio; break; +#endif + case 'C': configfile = optarg; break; + case 's': control_socket = optarg; break; default: fatal(0, "invalid option"); } } + if(config_read(0)) fatal(0, "cannot read configuration"); + if(!maxbuffer) + maxbuffer = 4 * readahead; argc -= optind; argv += optind; - if(argc < 1 || argc > 2) - fatal(0, "usage: disorder-playrtp [OPTIONS] ADDRESS [PORT]"); - sl.n = argc; - sl.s = argv; - /* Listen for inbound audio data */ - if(!(res = get_address(&sl, &pref, &sockname))) + switch(argc) { + case 0: + /* Get configuration from server */ + if(!(c = disorder_new(1))) exit(EXIT_FAILURE); + if(disorder_connect(c)) exit(EXIT_FAILURE); + if(disorder_rtp_address(c, &address, &port)) exit(EXIT_FAILURE); + sl.n = 2; + sl.s = xcalloc(2, sizeof *sl.s); + sl.s[0] = address; + sl.s[1] = port; + break; + case 1: + case 2: + /* Use command-line ADDRESS+PORT or just PORT */ + sl.n = argc; + sl.s = argv; + break; + default: + fatal(0, "usage: disorder-playrtp [OPTIONS] [[ADDRESS] PORT]"); + } + /* Look up address and port */ + if(!(res = get_address(&sl, &prefs, &sockname))) exit(1); + /* Create the socket */ if((rtpfd = socket(res->ai_family, res->ai_socktype, res->ai_protocol)) < 0) fatal(errno, "error creating socket"); + /* Stash the multicast group address */ + if((is_multicast = multicast(res->ai_addr))) { + memcpy(&mgroup, res->ai_addr, res->ai_addrlen); + switch(res->ai_addr->sa_family) { + case AF_INET: + mgroup.in.sin_port = 0; + break; + case AF_INET6: + mgroup.in6.sin6_port = 0; + break; + } + } + /* Bind to 0/port */ + switch(res->ai_addr->sa_family) { + case AF_INET: + memset(&((struct sockaddr_in *)res->ai_addr)->sin_addr, 0, + sizeof (struct in_addr)); + break; + case AF_INET6: + memset(&((struct sockaddr_in6 *)res->ai_addr)->sin6_addr, 0, + sizeof (struct in6_addr)); + break; + default: + fatal(0, "unsupported family %d", (int)res->ai_addr->sa_family); + } if(bind(rtpfd, res->ai_addr, res->ai_addrlen) < 0) fatal(errno, "error binding socket to %s", sockname); + if(is_multicast) { + switch(mgroup.sa.sa_family) { + case PF_INET: + mreq.imr_multiaddr = mgroup.in.sin_addr; + mreq.imr_interface.s_addr = 0; /* use primary interface */ + if(setsockopt(rtpfd, IPPROTO_IP, IP_ADD_MEMBERSHIP, + &mreq, sizeof mreq) < 0) + fatal(errno, "error calling setsockopt IP_ADD_MEMBERSHIP"); + break; + case PF_INET6: + mreq6.ipv6mr_multiaddr = mgroup.in6.sin6_addr; + memset(&mreq6.ipv6mr_interface, 0, sizeof mreq6.ipv6mr_interface); + if(setsockopt(rtpfd, IPPROTO_IPV6, IPV6_JOIN_GROUP, + &mreq6, sizeof mreq6) < 0) + fatal(errno, "error calling setsockopt IPV6_JOIN_GROUP"); + break; + default: + fatal(0, "unsupported address family %d", res->ai_family); + } + info("listening on %s multicast group %s", + format_sockaddr(res->ai_addr), format_sockaddr(&mgroup.sa)); + } else + info("listening on %s", format_sockaddr(res->ai_addr)); + len = sizeof rcvbuf; + if(getsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, &rcvbuf, &len) < 0) + fatal(errno, "error calling getsockopt SO_RCVBUF"); + if(target_rcvbuf > rcvbuf) { + if(setsockopt(rtpfd, SOL_SOCKET, SO_RCVBUF, + &target_rcvbuf, sizeof target_rcvbuf) < 0) + error(errno, "error calling setsockopt SO_RCVBUF %d", + target_rcvbuf); + /* We try to carry on anyway */ + else + info("changed socket receive buffer from %d to %d", + rcvbuf, target_rcvbuf); + } else + info("default socket receive buffer %d", rcvbuf); + if(logfp) + info("WARNING: -L option can impact performance"); + if(control_socket) { + pthread_t tid; + + if((err = pthread_create(&tid, 0, control_thread, 0))) + fatal(err, "pthread_create control_thread"); + } play_rtp(); return 0; }